| /* |
| * Copyright 2020 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #include <memory> |
| #include <optional> |
| #include <string> |
| #include <tuple> |
| #include <utility> |
| #include <vector> |
| |
| #include "absl/strings/string_view.h" |
| #include "api/environment/environment_factory.h" |
| #include "api/jsep.h" |
| #include "api/media_types.h" |
| #include "api/peer_connection_interface.h" |
| #include "api/rtc_error.h" |
| #include "api/rtc_event_log/rtc_event_log_factory.h" |
| #include "api/rtp_parameters.h" |
| #include "api/rtp_transceiver_direction.h" |
| #include "api/rtp_transceiver_interface.h" |
| #include "api/scoped_refptr.h" |
| #include "media/base/fake_media_engine.h" |
| #include "p2p/test/fake_port_allocator.h" |
| #include "pc/peer_connection_wrapper.h" |
| #include "pc/session_description.h" |
| #include "pc/test/enable_fake_media.h" |
| #include "pc/test/mock_peer_connection_observers.h" |
| #include "rtc_base/internal/default_socket_server.h" |
| #include "rtc_base/socket_server.h" |
| #include "rtc_base/strings/string_builder.h" |
| #include "rtc_base/thread.h" |
| #include "test/gmock.h" |
| #include "test/gtest.h" |
| |
| namespace webrtc { |
| |
| using ::testing::Combine; |
| using ::testing::ElementsAre; |
| using ::testing::Field; |
| using ::testing::Return; |
| using ::testing::Values; |
| |
| class PeerConnectionHeaderExtensionTest |
| : public ::testing::TestWithParam< |
| std::tuple<webrtc::MediaType, SdpSemantics>> { |
| protected: |
| PeerConnectionHeaderExtensionTest() |
| : socket_server_(CreateDefaultSocketServer()), |
| main_thread_(socket_server_.get()), |
| extensions_( |
| {RtpHeaderExtensionCapability("uri1", |
| 1, |
| RtpTransceiverDirection::kStopped), |
| RtpHeaderExtensionCapability("uri2", |
| 2, |
| RtpTransceiverDirection::kSendOnly), |
| RtpHeaderExtensionCapability("uri3", |
| 3, |
| RtpTransceiverDirection::kRecvOnly), |
| RtpHeaderExtensionCapability( |
| "uri4", |
| 4, |
| RtpTransceiverDirection::kSendRecv)}) {} |
| |
| std::unique_ptr<PeerConnectionWrapper> CreatePeerConnection( |
| webrtc::MediaType media_type, |
| std::optional<SdpSemantics> semantics) { |
| auto media_engine = std::make_unique<FakeMediaEngine>(); |
| if (media_type == webrtc::MediaType::AUDIO) |
| media_engine->fake_voice_engine()->SetRtpHeaderExtensions(extensions_); |
| else |
| media_engine->fake_video_engine()->SetRtpHeaderExtensions(extensions_); |
| PeerConnectionFactoryDependencies factory_dependencies; |
| factory_dependencies.network_thread = Thread::Current(); |
| factory_dependencies.worker_thread = Thread::Current(); |
| factory_dependencies.signaling_thread = Thread::Current(); |
| EnableFakeMedia(factory_dependencies, std::move(media_engine)); |
| |
| factory_dependencies.event_log_factory = |
| std::make_unique<RtcEventLogFactory>(); |
| |
| auto pc_factory = |
| CreateModularPeerConnectionFactory(std::move(factory_dependencies)); |
| |
| auto fake_port_allocator = std::make_unique<FakePortAllocator>( |
| CreateEnvironment(), socket_server_.get()); |
| auto observer = std::make_unique<MockPeerConnectionObserver>(); |
| PeerConnectionInterface::RTCConfiguration config; |
| if (semantics) |
| config.sdp_semantics = *semantics; |
| PeerConnectionDependencies pc_dependencies(observer.get()); |
| pc_dependencies.allocator = std::move(fake_port_allocator); |
| auto result = pc_factory->CreatePeerConnectionOrError( |
| config, std::move(pc_dependencies)); |
| EXPECT_TRUE(result.ok()); |
| observer->SetPeerConnectionInterface(result.value().get()); |
| return std::make_unique<PeerConnectionWrapper>( |
| pc_factory, result.MoveValue(), std::move(observer)); |
| } |
| |
| std::unique_ptr<SocketServer> socket_server_; |
| AutoSocketServerThread main_thread_; |
| std::vector<RtpHeaderExtensionCapability> extensions_; |
| }; |
| |
| TEST_P(PeerConnectionHeaderExtensionTest, TransceiverOffersHeaderExtensions) { |
| webrtc::MediaType media_type; |
| SdpSemantics semantics; |
| std::tie(media_type, semantics) = GetParam(); |
| if (semantics != SdpSemantics::kUnifiedPlan) |
| return; |
| std::unique_ptr<PeerConnectionWrapper> wrapper = |
| CreatePeerConnection(media_type, semantics); |
| auto transceiver = wrapper->AddTransceiver(media_type); |
| EXPECT_EQ(transceiver->GetHeaderExtensionsToNegotiate(), extensions_); |
| } |
| |
| TEST_P(PeerConnectionHeaderExtensionTest, |
| SenderReceiverCapabilitiesReturnNotStoppedExtensions) { |
| webrtc::MediaType media_type; |
| SdpSemantics semantics; |
| std::tie(media_type, semantics) = GetParam(); |
| std::unique_ptr<PeerConnectionWrapper> wrapper = |
| CreatePeerConnection(media_type, semantics); |
| EXPECT_THAT(wrapper->pc_factory() |
| ->GetRtpSenderCapabilities(media_type) |
| .header_extensions, |
| ElementsAre(Field(&RtpHeaderExtensionCapability::uri, "uri2"), |
| Field(&RtpHeaderExtensionCapability::uri, "uri3"), |
| Field(&RtpHeaderExtensionCapability::uri, "uri4"))); |
| EXPECT_EQ(wrapper->pc_factory() |
| ->GetRtpReceiverCapabilities(media_type) |
| .header_extensions, |
| wrapper->pc_factory() |
| ->GetRtpSenderCapabilities(media_type) |
| .header_extensions); |
| } |
| |
| TEST_P(PeerConnectionHeaderExtensionTest, OffersUnstoppedDefaultExtensions) { |
| webrtc::MediaType media_type; |
| SdpSemantics semantics; |
| std::tie(media_type, semantics) = GetParam(); |
| if (semantics != SdpSemantics::kUnifiedPlan) |
| return; |
| std::unique_ptr<PeerConnectionWrapper> wrapper = |
| CreatePeerConnection(media_type, semantics); |
| auto transceiver = wrapper->AddTransceiver(media_type); |
| auto session_description = wrapper->CreateOffer(); |
| EXPECT_THAT(session_description->description() |
| ->contents()[0] |
| .media_description() |
| ->rtp_header_extensions(), |
| ElementsAre(Field(&RtpExtension::uri, "uri2"), |
| Field(&RtpExtension::uri, "uri3"), |
| Field(&RtpExtension::uri, "uri4"))); |
| } |
| |
| TEST_P(PeerConnectionHeaderExtensionTest, OffersUnstoppedModifiedExtensions) { |
| webrtc::MediaType media_type; |
| SdpSemantics semantics; |
| std::tie(media_type, semantics) = GetParam(); |
| if (semantics != SdpSemantics::kUnifiedPlan) |
| return; |
| std::unique_ptr<PeerConnectionWrapper> wrapper = |
| CreatePeerConnection(media_type, semantics); |
| auto transceiver = wrapper->AddTransceiver(media_type); |
| auto modified_extensions = transceiver->GetHeaderExtensionsToNegotiate(); |
| modified_extensions[0].direction = RtpTransceiverDirection::kSendRecv; |
| modified_extensions[3].direction = RtpTransceiverDirection::kStopped; |
| EXPECT_TRUE( |
| transceiver->SetHeaderExtensionsToNegotiate(modified_extensions).ok()); |
| auto session_description = wrapper->CreateOffer(); |
| EXPECT_THAT(session_description->description() |
| ->contents()[0] |
| .media_description() |
| ->rtp_header_extensions(), |
| ElementsAre(Field(&RtpExtension::uri, "uri1"), |
| Field(&RtpExtension::uri, "uri2"), |
| Field(&RtpExtension::uri, "uri3"))); |
| } |
| |
| TEST_P(PeerConnectionHeaderExtensionTest, AnswersUnstoppedModifiedExtensions) { |
| webrtc::MediaType media_type; |
| SdpSemantics semantics; |
| std::tie(media_type, semantics) = GetParam(); |
| if (semantics != SdpSemantics::kUnifiedPlan) |
| return; |
| std::unique_ptr<PeerConnectionWrapper> pc1 = |
| CreatePeerConnection(media_type, semantics); |
| std::unique_ptr<PeerConnectionWrapper> pc2 = |
| CreatePeerConnection(media_type, semantics); |
| auto transceiver1 = pc1->AddTransceiver(media_type); |
| |
| auto offer = pc1->CreateOfferAndSetAsLocal( |
| PeerConnectionInterface::RTCOfferAnswerOptions()); |
| pc2->SetRemoteDescription(std::move(offer)); |
| |
| ASSERT_EQ(pc2->pc()->GetTransceivers().size(), 1u); |
| auto transceiver2 = pc2->pc()->GetTransceivers()[0]; |
| auto modified_extensions = transceiver2->GetHeaderExtensionsToNegotiate(); |
| // Don't offer uri4. |
| modified_extensions[3].direction = RtpTransceiverDirection::kStopped; |
| transceiver2->SetHeaderExtensionsToNegotiate(modified_extensions); |
| |
| auto answer = pc2->CreateAnswerAndSetAsLocal( |
| PeerConnectionInterface::RTCOfferAnswerOptions()); |
| EXPECT_THAT(answer->description() |
| ->contents()[0] |
| .media_description() |
| ->rtp_header_extensions(), |
| ElementsAre(Field(&RtpExtension::uri, "uri2"), |
| Field(&RtpExtension::uri, "uri3"))); |
| } |
| |
| TEST_P(PeerConnectionHeaderExtensionTest, NegotiatedExtensionsAreAccessible) { |
| webrtc::MediaType media_type; |
| SdpSemantics semantics; |
| std::tie(media_type, semantics) = GetParam(); |
| if (semantics != SdpSemantics::kUnifiedPlan) |
| return; |
| std::unique_ptr<PeerConnectionWrapper> pc1 = |
| CreatePeerConnection(media_type, semantics); |
| auto transceiver1 = pc1->AddTransceiver(media_type); |
| auto modified_extensions = transceiver1->GetHeaderExtensionsToNegotiate(); |
| modified_extensions[3].direction = RtpTransceiverDirection::kStopped; |
| transceiver1->SetHeaderExtensionsToNegotiate(modified_extensions); |
| auto offer = pc1->CreateOfferAndSetAsLocal( |
| PeerConnectionInterface::RTCOfferAnswerOptions()); |
| |
| std::unique_ptr<PeerConnectionWrapper> pc2 = |
| CreatePeerConnection(media_type, semantics); |
| auto transceiver2 = pc2->AddTransceiver(media_type); |
| pc2->SetRemoteDescription(std::move(offer)); |
| auto answer = pc2->CreateAnswerAndSetAsLocal( |
| PeerConnectionInterface::RTCOfferAnswerOptions()); |
| pc1->SetRemoteDescription(std::move(answer)); |
| |
| // PC1 has exts 2-4 unstopped and PC2 has exts 1-3 unstopped -> ext 2, 3 |
| // survives. |
| EXPECT_THAT(transceiver1->GetNegotiatedHeaderExtensions(), |
| ElementsAre(Field(&RtpHeaderExtensionCapability::direction, |
| RtpTransceiverDirection::kStopped), |
| Field(&RtpHeaderExtensionCapability::direction, |
| RtpTransceiverDirection::kSendRecv), |
| Field(&RtpHeaderExtensionCapability::direction, |
| RtpTransceiverDirection::kSendRecv), |
| Field(&RtpHeaderExtensionCapability::direction, |
| RtpTransceiverDirection::kStopped))); |
| } |
| |
| TEST_P(PeerConnectionHeaderExtensionTest, OfferedExtensionsArePerTransceiver) { |
| webrtc::MediaType media_type; |
| SdpSemantics semantics; |
| std::tie(media_type, semantics) = GetParam(); |
| if (semantics != SdpSemantics::kUnifiedPlan) |
| return; |
| std::unique_ptr<PeerConnectionWrapper> pc1 = |
| CreatePeerConnection(media_type, semantics); |
| auto transceiver1 = pc1->AddTransceiver(media_type); |
| auto modified_extensions = transceiver1->GetHeaderExtensionsToNegotiate(); |
| modified_extensions[3].direction = RtpTransceiverDirection::kStopped; |
| transceiver1->SetHeaderExtensionsToNegotiate(modified_extensions); |
| auto transceiver2 = pc1->AddTransceiver(media_type); |
| |
| auto session_description = pc1->CreateOffer(); |
| EXPECT_THAT(session_description->description() |
| ->contents()[0] |
| .media_description() |
| ->rtp_header_extensions(), |
| ElementsAre(Field(&RtpExtension::uri, "uri2"), |
| Field(&RtpExtension::uri, "uri3"))); |
| EXPECT_THAT(session_description->description() |
| ->contents()[1] |
| .media_description() |
| ->rtp_header_extensions(), |
| ElementsAre(Field(&RtpExtension::uri, "uri2"), |
| Field(&RtpExtension::uri, "uri3"), |
| Field(&RtpExtension::uri, "uri4"))); |
| } |
| |
| TEST_P(PeerConnectionHeaderExtensionTest, RemovalAfterRenegotiation) { |
| webrtc::MediaType media_type; |
| SdpSemantics semantics; |
| std::tie(media_type, semantics) = GetParam(); |
| if (semantics != SdpSemantics::kUnifiedPlan) |
| return; |
| std::unique_ptr<PeerConnectionWrapper> pc1 = |
| CreatePeerConnection(media_type, semantics); |
| std::unique_ptr<PeerConnectionWrapper> pc2 = |
| CreatePeerConnection(media_type, semantics); |
| auto transceiver1 = pc1->AddTransceiver(media_type); |
| |
| auto offer = pc1->CreateOfferAndSetAsLocal( |
| PeerConnectionInterface::RTCOfferAnswerOptions()); |
| pc2->SetRemoteDescription(std::move(offer)); |
| auto answer = pc2->CreateAnswerAndSetAsLocal( |
| PeerConnectionInterface::RTCOfferAnswerOptions()); |
| pc1->SetRemoteDescription(std::move(answer)); |
| |
| auto modified_extensions = transceiver1->GetHeaderExtensionsToNegotiate(); |
| modified_extensions[3].direction = RtpTransceiverDirection::kStopped; |
| transceiver1->SetHeaderExtensionsToNegotiate(modified_extensions); |
| auto session_description = pc1->CreateOffer(); |
| EXPECT_THAT(session_description->description() |
| ->contents()[0] |
| .media_description() |
| ->rtp_header_extensions(), |
| ElementsAre(Field(&RtpExtension::uri, "uri2"), |
| Field(&RtpExtension::uri, "uri3"))); |
| } |
| |
| TEST_P(PeerConnectionHeaderExtensionTest, |
| StoppedByDefaultExtensionCanBeActivatedByRemoteSdp) { |
| webrtc::MediaType media_type; |
| SdpSemantics semantics; |
| std::tie(media_type, semantics) = GetParam(); |
| if (semantics != SdpSemantics::kUnifiedPlan) |
| return; |
| std::unique_ptr<PeerConnectionWrapper> pc1 = |
| CreatePeerConnection(media_type, semantics); |
| std::unique_ptr<PeerConnectionWrapper> pc2 = |
| CreatePeerConnection(media_type, semantics); |
| auto transceiver1 = pc1->AddTransceiver(media_type); |
| |
| auto offer = pc1->CreateOfferAndSetAsLocal( |
| PeerConnectionInterface::RTCOfferAnswerOptions()); |
| pc2->SetRemoteDescription(std::move(offer)); |
| auto answer = pc2->CreateAnswerAndSetAsLocal( |
| PeerConnectionInterface::RTCOfferAnswerOptions()); |
| std::string sdp; |
| ASSERT_TRUE(answer->ToString(&sdp)); |
| // We support uri1 but it is stopped by default. Let the remote reactivate it. |
| sdp += "a=extmap:15 uri1\r\n"; |
| auto modified_answer = CreateSessionDescription(SdpType::kAnswer, sdp); |
| pc1->SetRemoteDescription(std::move(modified_answer)); |
| EXPECT_THAT(transceiver1->GetNegotiatedHeaderExtensions(), |
| ElementsAre(Field(&RtpHeaderExtensionCapability::direction, |
| RtpTransceiverDirection::kSendRecv), |
| Field(&RtpHeaderExtensionCapability::direction, |
| RtpTransceiverDirection::kSendRecv), |
| Field(&RtpHeaderExtensionCapability::direction, |
| RtpTransceiverDirection::kSendRecv), |
| Field(&RtpHeaderExtensionCapability::direction, |
| RtpTransceiverDirection::kSendRecv))); |
| } |
| |
| TEST_P(PeerConnectionHeaderExtensionTest, |
| UnknownExtensionInRemoteOfferDoesNotShowUp) { |
| webrtc::MediaType media_type; |
| SdpSemantics semantics; |
| std::tie(media_type, semantics) = GetParam(); |
| if (semantics != SdpSemantics::kUnifiedPlan) |
| return; |
| std::unique_ptr<PeerConnectionWrapper> pc = |
| CreatePeerConnection(media_type, semantics); |
| std::string sdp = |
| "v=0\r\n" |
| "o=- 0 3 IN IP4 127.0.0.1\r\n" |
| "s=-\r\n" |
| "t=0 0\r\n" |
| "a=fingerprint:sha-256 " |
| "A7:24:72:CA:6E:02:55:39:BA:66:DF:6E:CC:4C:D8:B0:1A:BF:1A:56:65:7D:F4:03:" |
| "AD:7E:77:43:2A:29:EC:93\r\n" |
| "a=ice-ufrag:6HHHdzzeIhkE0CKj\r\n" |
| "a=ice-pwd:XYDGVpfvklQIEnZ6YnyLsAew\r\n"; |
| if (media_type == webrtc::MediaType::AUDIO) { |
| sdp += |
| "m=audio 9 RTP/AVPF 111\r\n" |
| "a=rtpmap:111 fake_audio_codec/8000\r\n"; |
| } else { |
| sdp += |
| "m=video 9 RTP/AVPF 111\r\n" |
| "a=rtpmap:111 fake_video_codec/90000\r\n"; |
| } |
| sdp += |
| "c=IN IP4 0.0.0.0\r\n" |
| "a=rtcp-mux\r\n" |
| "a=sendonly\r\n" |
| "a=mid:audio\r\n" |
| "a=setup:actpass\r\n" |
| "a=extmap:1 urn:bogus\r\n"; |
| auto offer = CreateSessionDescription(SdpType::kOffer, sdp); |
| pc->SetRemoteDescription(std::move(offer)); |
| pc->CreateAnswerAndSetAsLocal( |
| PeerConnectionInterface::RTCOfferAnswerOptions()); |
| ASSERT_GT(pc->pc()->GetTransceivers().size(), 0u); |
| auto transceiver = pc->pc()->GetTransceivers()[0]; |
| auto negotiated = transceiver->GetNegotiatedHeaderExtensions(); |
| EXPECT_EQ(negotiated.size(), |
| transceiver->GetHeaderExtensionsToNegotiate().size()); |
| // All extensions are stopped, the "bogus" one does not show up. |
| for (const auto& extension : negotiated) { |
| EXPECT_EQ(extension.direction, RtpTransceiverDirection::kStopped); |
| EXPECT_NE(extension.uri, "urn:bogus"); |
| } |
| } |
| |
| // These tests are regression tests for behavior that the API |
| // enables in a proper way. It conflicts with the behavior |
| // of the API to only offer non-stopped extensions. |
| TEST_P(PeerConnectionHeaderExtensionTest, |
| SdpMungingAnswerWithoutApiUsageEnablesExtensions) { |
| webrtc::MediaType media_type; |
| SdpSemantics semantics; |
| std::tie(media_type, semantics) = GetParam(); |
| if (semantics != SdpSemantics::kUnifiedPlan) |
| return; |
| std::unique_ptr<PeerConnectionWrapper> pc = |
| CreatePeerConnection(media_type, semantics); |
| std::string sdp = |
| "v=0\r\n" |
| "o=- 0 3 IN IP4 127.0.0.1\r\n" |
| "s=-\r\n" |
| "t=0 0\r\n" |
| "a=fingerprint:sha-256 " |
| "A7:24:72:CA:6E:02:55:39:BA:66:DF:6E:CC:4C:D8:B0:1A:BF:1A:56:65:7D:F4:03:" |
| "AD:7E:77:43:2A:29:EC:93\r\n" |
| "a=ice-ufrag:6HHHdzzeIhkE0CKj\r\n" |
| "a=ice-pwd:XYDGVpfvklQIEnZ6YnyLsAew\r\n"; |
| if (media_type == webrtc::MediaType::AUDIO) { |
| sdp += |
| "m=audio 9 RTP/AVPF 111\r\n" |
| "a=rtpmap:111 fake_audio_codec/8000\r\n"; |
| } else { |
| sdp += |
| "m=video 9 RTP/AVPF 111\r\n" |
| "a=rtpmap:111 fake_video_codec/90000\r\n"; |
| } |
| sdp += |
| "c=IN IP4 0.0.0.0\r\n" |
| "a=rtcp-mux\r\n" |
| "a=sendrecv\r\n" |
| "a=mid:audio\r\n" |
| "a=setup:actpass\r\n" |
| "a=extmap:1 uri1\r\n"; |
| auto offer = CreateSessionDescription(SdpType::kOffer, sdp); |
| pc->SetRemoteDescription(std::move(offer)); |
| auto answer = |
| pc->CreateAnswer(PeerConnectionInterface::RTCOfferAnswerOptions()); |
| std::string modified_sdp; |
| ASSERT_TRUE(answer->ToString(&modified_sdp)); |
| modified_sdp += "a=extmap:1 uri1\r\n"; |
| auto modified_answer = |
| CreateSessionDescription(SdpType::kAnswer, modified_sdp); |
| ASSERT_TRUE(pc->SetLocalDescription(std::move(modified_answer))); |
| |
| auto session_description = pc->CreateOffer(); |
| EXPECT_THAT(session_description->description() |
| ->contents()[0] |
| .media_description() |
| ->rtp_header_extensions(), |
| ElementsAre(Field(&RtpExtension::uri, "uri1"), |
| Field(&RtpExtension::uri, "uri2"), |
| Field(&RtpExtension::uri, "uri3"), |
| Field(&RtpExtension::uri, "uri4"))); |
| } |
| |
| TEST_P(PeerConnectionHeaderExtensionTest, |
| SdpMungingOfferWithoutApiUsageEnablesExtensions) { |
| webrtc::MediaType media_type; |
| SdpSemantics semantics; |
| std::tie(media_type, semantics) = GetParam(); |
| if (semantics != SdpSemantics::kUnifiedPlan) |
| return; |
| std::unique_ptr<PeerConnectionWrapper> pc = |
| CreatePeerConnection(media_type, semantics); |
| pc->AddTransceiver(media_type); |
| |
| auto offer = |
| pc->CreateOffer(PeerConnectionInterface::RTCOfferAnswerOptions()); |
| std::string modified_sdp; |
| ASSERT_TRUE(offer->ToString(&modified_sdp)); |
| modified_sdp += "a=extmap:1 uri1\r\n"; |
| auto modified_offer = CreateSessionDescription(SdpType::kOffer, modified_sdp); |
| ASSERT_TRUE(pc->SetLocalDescription(std::move(modified_offer))); |
| |
| auto offer2 = |
| pc->CreateOffer(PeerConnectionInterface::RTCOfferAnswerOptions()); |
| EXPECT_THAT(offer2->description() |
| ->contents()[0] |
| .media_description() |
| ->rtp_header_extensions(), |
| ElementsAre(Field(&RtpExtension::uri, "uri2"), |
| Field(&RtpExtension::uri, "uri3"), |
| Field(&RtpExtension::uri, "uri4"), |
| Field(&RtpExtension::uri, "uri1"))); |
| } |
| |
| TEST_P(PeerConnectionHeaderExtensionTest, EnablingExtensionsAfterRemoteOffer) { |
| webrtc::MediaType media_type; |
| SdpSemantics semantics; |
| std::tie(media_type, semantics) = GetParam(); |
| if (semantics != SdpSemantics::kUnifiedPlan) |
| return; |
| std::unique_ptr<PeerConnectionWrapper> pc = |
| CreatePeerConnection(media_type, semantics); |
| std::string sdp = |
| "v=0\r\n" |
| "o=- 0 3 IN IP4 127.0.0.1\r\n" |
| "s=-\r\n" |
| "t=0 0\r\n" |
| "a=fingerprint:sha-256 " |
| "A7:24:72:CA:6E:02:55:39:BA:66:DF:6E:CC:4C:D8:B0:1A:BF:1A:56:65:7D:F4:03:" |
| "AD:7E:77:43:2A:29:EC:93\r\n" |
| "a=ice-ufrag:6HHHdzzeIhkE0CKj\r\n" |
| "a=ice-pwd:XYDGVpfvklQIEnZ6YnyLsAew\r\n"; |
| if (media_type == webrtc::MediaType::AUDIO) { |
| sdp += |
| "m=audio 9 RTP/AVPF 111\r\n" |
| "a=rtpmap:111 fake_audio_codec/8000\r\n"; |
| } else { |
| sdp += |
| "m=video 9 RTP/AVPF 111\r\n" |
| "a=rtpmap:111 fake_video_codec/90000\r\n"; |
| } |
| sdp += |
| "c=IN IP4 0.0.0.0\r\n" |
| "a=rtcp-mux\r\n" |
| "a=sendrecv\r\n" |
| "a=mid:audio\r\n" |
| "a=setup:actpass\r\n" |
| "a=extmap:5 uri1\r\n"; |
| auto offer = CreateSessionDescription(SdpType::kOffer, sdp); |
| pc->SetRemoteDescription(std::move(offer)); |
| |
| ASSERT_GT(pc->pc()->GetTransceivers().size(), 0u); |
| auto transceiver = pc->pc()->GetTransceivers()[0]; |
| auto modified_extensions = transceiver->GetHeaderExtensionsToNegotiate(); |
| modified_extensions[0].direction = RtpTransceiverDirection::kSendRecv; |
| transceiver->SetHeaderExtensionsToNegotiate(modified_extensions); |
| |
| pc->CreateAnswerAndSetAsLocal( |
| PeerConnectionInterface::RTCOfferAnswerOptions()); |
| |
| auto session_description = pc->CreateOffer(); |
| auto extensions = session_description->description() |
| ->contents()[0] |
| .media_description() |
| ->rtp_header_extensions(); |
| EXPECT_THAT(extensions, ElementsAre(Field(&RtpExtension::uri, "uri1"), |
| Field(&RtpExtension::uri, "uri2"), |
| Field(&RtpExtension::uri, "uri3"), |
| Field(&RtpExtension::uri, "uri4"))); |
| // Check uri1's id still matches the remote id. |
| EXPECT_EQ(extensions[0].id, 5); |
| } |
| |
| INSTANTIATE_TEST_SUITE_P( |
| , |
| PeerConnectionHeaderExtensionTest, |
| Combine(Values(SdpSemantics::kPlanB_DEPRECATED, SdpSemantics::kUnifiedPlan), |
| Values(webrtc::MediaType::AUDIO, webrtc::MediaType::VIDEO)), |
| [](const testing::TestParamInfo< |
| PeerConnectionHeaderExtensionTest::ParamType>& info) { |
| webrtc::MediaType media_type; |
| SdpSemantics semantics; |
| std::tie(media_type, semantics) = info.param; |
| return (StringBuilder("With") |
| << (semantics == SdpSemantics::kPlanB_DEPRECATED ? "PlanB" |
| : "UnifiedPlan") |
| << "And" |
| << (media_type == webrtc::MediaType::AUDIO ? "Voice" : "Video") |
| << "Engine") |
| .str(); |
| }); |
| |
| } // namespace webrtc |
